Modern digital networks are made to operate in a multimedia environment for transporting different types of data, i.e. pure data or digitized information signals (including voice, image, video, etc . . . ) over the same high speed digital network, while ensuring the compliance with the requirements specific to each kind of these data traffics.
Transporting over same network means that in practice, multimedia sources are physically connected to same network nodes and multiplexed over same network links (also called trunks) interconnecting the network nodes. Thus the different types of traffic share the same links/trunks bandwidths. To that end, a so called packet switching technique is used whereby the digitized data are arranged into so called bit packets, which packets are then multiplexed over the same links with their specific bandwidth being assigned according to predefined bandwidth criteria.
One may notice that the information provided by the various users can be divided into different types. These include non-real-time type of information, i.e. information that can be delivered to a destination end user of the network with minor time constraint restrictions; and real time type information that must be transmitted to the end-user with a predefined limited delay restriction The latter type includes voice information If some real time type information is not transferred within said time delay, it should simply be discarded, bearing in mind however that techniques have been provided for recovering, to some extent, the discarded data These techniques include so-called interpolation/extrapolation These techniques do however have their limitations
On the other hand, non real time data traffic can be delayed but suffer no loss. Techniques have been developed accordingly for managing the corresponding data traffic. These techniques include retry and re-transmissions that, unfortunately add to the data traffic which is already heavy to manage and support.
Bearing in mind the above requirements and characteristics of the various types of traffics, one may understand that while optimizing the available transmission bandwidth is a must, it is however far from being simple to achieve This is even emphasized when considering the network architectures presently available in the field.
Cost efficiency requirements lead to designing communication networks that have at their disposal limited resources at all network levels and more particularly at the level of link/trunks bandwidths and node resources to manage these bandwidths and control the traffic.
It is to be noted that in the following description the expression link or trunk shall be used equally with no difference being made, as far as this invention is concerned.
As per the transport of voice originated data, it has already been noted that these data do appear at the source level in so-called "talkspurts" form with long silences in between Accordingly, techniques have already been used to optimize the assigned communication bandwidth For instance, a so called TASI Technique has been used for concentrating a very large number of sources into a smaller number of channels This technique is based on statistical considerations which do apply to a very large numbers of sources concentrated on a same network entry But this is not the case with actual high speed digital networks processing multimedia information, in which one or a small number of PBXs or PABXs, herein designated as audio type sources, might be connected to a network entry node, together with other data sources such as host computers, servers, video sources, etc . . .
There is, therefore, a need for developing methods and means for optimizing the communication bandwidth assigned/reserved to audio type sources (i.e. assigned to voice type channels) and efficiently manage the network links bandwidth accordingly.